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gerrit-public.fairphone.software
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platform
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external
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webrtc
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08b0351ddd0469c8b0aa7c028ac93878a350e593
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869dab7
Disable Intel VP8 HW encoder.
by Alex Glaznev
· 8 years ago
6a35590
Add code for dummy file audio to fallback to dummy audio.
by noahric
· 8 years ago
7c0f8ee
Avoid null pointer exception if Android getCameraInfo fails.
by Alex Glaznev
· 8 years ago
d8a72f0
Close input file in FileAudioDevice::StopRecording.
by noahric
· 8 years ago
78810b6
Expose media constraint string constants as ObjC NSStrings
by magjed
· 8 years ago
d22854b
FilePlayer: Remove unused default values for arguments
by kwiberg
· 8 years ago
4a42900
Removes redundant log warning in WebRtcAudioManager.
by henrika
· 8 years ago
86c9694
Revert of StartTimestamp generated randomly in RtpSender constructor (patchset #4 id:60001 of https://codereview.webrtc.org/2241193002/ )
by danilchap
· 8 years ago
5a25d95
FileRecorder + FilePlayer: Let Create functions return unique_ptr
by kwiberg
· 8 years ago
4466782
StartTimestamp generated randomly in RtpSender constructor
by Danil Chapovalov
· 8 years ago
2ae1fb6
Fix get_landmines.py script.
by ehmaldonado
· 8 years ago
144dd27
FileRecorderImpl and FilePlayerImpl don't need their own .h and .cc files
by kwiberg
· 8 years ago
c54071d
WebRtcVoiceEngine: Use AudioDecoderFactory to generate recv codecs.
by ossu
· 8 years ago
a93d5ac
Don't simulate probing based on rtc event logs since we don't have that info logged.
by stefan
· 8 years ago
eb680ea
CongestionController::SetBweBitrates may now trigger probing.
by philipel
· 8 years ago
c594aa61
Add a gyp/gn option to use dummy audio file devices.
by noahric
· 8 years ago
e05bcc2
Do not switch a connection if the new connection is not ready to send packets.
by Honghai Zhang
· 8 years ago
49c01d7
Currently there is not way to programmically test whether a ScreenCapturer
by zijiehe
· 8 years ago
895e1a9
Change the default backup connection ping interval to 25 seconds.
by Honghai Zhang
· 8 years ago
287e548
Cleanup RtcpReceiver::TMMBRReceived function
by danilchap
· 8 years ago
f095012
Revert of Adding audio to video_quality_test. (patchset #10 id:230001 of https://codereview.webrtc.org/2136573002/ )
by minyue
· 8 years ago
65a6578
Adding audio to video_quality_test.
by minyue
· 8 years ago
75c287e
Fix incorrect example in mod_ops.h
by philipel
· 8 years ago
a06ce49
Run "git cl format" on some files before I start to modify them
by kwiberg
· 8 years ago
b789439
Roll chromium_revision 2b53ee0889..915e47250f (411979:412201)
by buildbot
· 8 years ago
90920d5
GN: Enable msse2 flag in Mac.
by ehmaldonado
· 8 years ago
9d7eb13
Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #3 id:40001 of https://codereview.webrtc.org/2247033003/ )
by kwiberg
· 8 years ago
e252d3c
MB: Fix incorrect iOS builder names.
by kjellander
· 8 years ago
427ce3d
Move FilePlayer and FileRecorder to Voice Engine
by kwiberg
· 8 years ago
2f69ce9
Cleaned out candidateSet member from TMMBRHelp class
by danilchap
· 8 years ago
1c814e7
iOS: Update MB and JSON configs + enable Goma
by kjellander
· 8 years ago
8eb37a3
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
by perkj
· 8 years ago
6910537
Add gn target for audio_device_tests.
by maxmorin
· 8 years ago
70f866c
Added new mixer to |check_targets| in .gn and fixed include/depend errors.
by aleloi
· 8 years ago
7522a28
Removed old probe cluster logic and logic related to ssrcs from DelayBasedBwe.
by philipel
· 8 years ago
b7186d0
Migrated GN target :isac_fix_test
by aleloi
· 8 years ago
b24b1ce
Moving mock classes around so that they may be reused in other unittests
by hbos
· 8 years ago
88e31a3
Fix warnings, simplify ADM.
by maxmorin
· 8 years ago
cc16836
- Add task queue to Call with the intent of replacing the use of one of the process threads.
by perkj
· 8 years ago
82dda1a
[WebRTC] Disable DirectX capturer tests if the system does not support it.
by zijiehe
· 8 years ago
e1b4d24
Skip AUD while extracting SPS and PPS on iOS.
by jianjun.zhu
· 8 years ago
6c687e7
Make prior H264 QP adjustments iOS specific.
by tkchin
· 8 years ago
3473288
Remove VERBOSE logs in (android) audio device code.
by noahric
· 8 years ago
43ba317
Roll chromium_revision 4b42aa218b..2b53ee0889 (411951:411979)
by buildbot
· 8 years ago
b1e6611
GN: Fix audio_decoder_unittests for android.
by ehmaldonado
· 8 years ago
4a1ec1e
Added ProbeBitrate(bitrate_bps, num_probes) to BitrateProber.
by philipel
· 8 years ago
1aee0b5
Remove old methods in AudioTransport, make it pass a gn build
by maxmorin
· 8 years ago
c8c71f4
Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #6 id:100001 of https://codereview.webrtc.org/2240163002/ )
by kwiberg
· 8 years ago
dc65ea2
Move FilePlayer and FileRecorder to Voice Engine
by kwiberg
· 8 years ago
f96c51a
GN: Add video_capture_tests for Mac
by kjellander
· 8 years ago
2a75801
Revert of CQ: Temporarily disable iOS Simulator trybots (patchset #1 id:1 of https://codereview.webrtc.org/2244183002/ )
by kjellander
· 8 years ago
e34c19c
Clarify some function names in visualization tool.
by terelius
· 8 years ago
2ab1da7
Revert of Added new mixer to |check_targets| in .gn and fixed include/depend errors. (patchset #1 id:1 of https://codereview.webrtc.org/2234293002/ )
by olka
· 8 years ago
d700bef
Added new mixer to |check_targets| in .gn and fixed include/depend errors.
by aleloi
· 8 years ago
963be23
RtpRtcp: Remove the SetSendREDPayloadType and SendREDPayloadType methods
by kwiberg
· 8 years ago
8f956de
FakeTiming added, an implementation of Timing that can be used for tests.
by hbos
· 8 years ago
96dbc8f
Adding comment regarding the disabling the flaky test VolumeTest.ManualRequiresMicrophoneCanSetMicrophoneVolumeWithAgcOff
by peah
· 8 years ago
3ab6614
Add video_loopback to gn.
by stefan
· 8 years ago
92c0950
Make CameraCapturer.switchCamera try again if session is still opening.
by sakal
· 8 years ago
d7d05f8
Disabling the test VolumeTest.ManualRequiresMicrophoneCanSetMicrophoneVolumeWithAgcOff
by peah
· 8 years ago
da07af2
Roll chromium_revision 7f405ec2b6..4b42aa218b (411933:411951)
by buildbot
· 8 years ago
3e3ebe6
remove unnecessary double allocation
by kthelgason
· 8 years ago
0ccff57
VoERTP_RTCP: Remove GetREDStatus and SetREDStatus
by kwiberg
· 8 years ago
5bcc00e
Changed folder structure in new mixer and fixed simple lint errors.
by aleloi
· 8 years ago
714dd4e
GN: Update tests to have the correct shard timeout value on Android.
by sakal
· 8 years ago
5093b38
Make variable for selecting if intervals without samples should be included in stats configurable (for rate counters).
by asapersson
· 8 years ago
c61ae74
Roll chromium_revision 941118827f..7f405ec2b6 (411223:411933)
by buildbot
· 8 years ago
414eb18
CQ: Temporarily disable iOS Simulator trybots
by kjellander
· 8 years ago
4cb5b64
Fix for data channels perpetually stuck in "closing" state.
by Taylor Brandstetter
· 8 years ago
64a7eab
Update tests and DTX check for Opus 1.1.3.
by flim
· 8 years ago
9591e3e
Convert PeerConnectionTest to use the new capture APIs.
by sakal
· 8 years ago
62351c9
Fixing problems with ICE candidate pair prioritization.
by Taylor Brandstetter
· 8 years ago
6f82535
Enabling IPv6 socket recv timestamp test, and making less flaky.
by Taylor Brandstetter
· 8 years ago
588783a
Return nil from RTCPeerConnectionFactory when creation fails
by skvlad
· 8 years ago
fe1ffb1
Remove unused SessionId from TransportChannel and PortAllocatorSession.
by johan
· 8 years ago
c8762a8
Remove StartSSLWithServer from SSLStreamAdapter.
by Taylor Brandstetter
· 8 years ago
f10976e
Roll chromium_revision db8d32de07..941118827f (410624:411223)
by kjellander
· 8 years ago
3b74768
Remove pbos@webrtc.org from WATCHLISTS.
by Peter Boström
· 8 years ago
2e5cfcd
Add periodic logging of video stats.
by asapersson
· 8 years ago
b179767
Add an HD resolution perf test.
by stefan
· 8 years ago
17deeb4
PacketBuffer is now ref counted.
by philipel
· 8 years ago
a3a1fde
Add Mac bots to MB.
by ehmaldonado
· 8 years ago
d30e0ad
Session based capturing for Camera2Capturer.
by sakal
· 8 years ago
bd59c71
GN: Add dependency libjingle_peerconnection_java to modules_unittests.
by sakal
· 8 years ago
0ae7878
MB: Add Windows configurations
by kjellander
· 8 years ago
3d31bd6
Do not create incompatible TurnPort if the server address family is known.
by Honghai Zhang
· 8 years ago
bf8a2c9
Probe bitrate estimator correction.
by philipel
· 8 years ago
68815bf
MB: Make all Android debug builds static
by kjellander
· 8 years ago
c99d5a6
Add stefan@ to webrtc/OWNERS.
by stefan
· 8 years ago
63cb172
MB: Fix typo for android_arm64_rel trybot.
by kjellander
· 8 years ago
fb372f0
iOS render: Handle frame rotation in OpenGL
by magjed
· 8 years ago
4556b45
Fix tools_unittests in GN.
by ehmaldonado
· 8 years ago
ccbbf8d
Visualize delay changes based on both abs-send-time and capture time.
by terelius
· 8 years ago
d49a37b
Rename main file for visualization tool.
by terelius
· 8 years ago
c4ac700
Migrated GN target :neteq_pcmu_quality_test
by aleloi
· 8 years ago
6df36dc
Migrated GN target :neteq_isac_quality_test
by aleloi
· 8 years ago
e6ca9ec
Broke out 'level_indicator' of the voice_engine GN target. This is
by aleloi
· 8 years ago
0e0be0a
Migrated GN target :neteq_ilbc_quality_test
by aleloi
· 8 years ago
6e6e70f
Add magjed@webrtc.org as owner of webrtc/base/java/
by Magnus Jedvert
· 8 years ago
6391012
Migrated GN target :audio_classifier_test
by aleloi
· 8 years ago
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